View source for Internet of Things and VoIP

=Introduction=
This topic is talking about to set up the relationship between IoT and IP Telephony network. 

We will do different tests below to show how to integrate IoT into VoIP network. We will use the open source [http://www.asterisk.org/ Asterisk] software here for voip connection. 

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=Test 1: Install Asterisk in Dragino make internal calls=
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==Purpose==
Asterisk is a powerful, open source software to build telephony system. Softphone can be runed in either PC or smart phone. The purpose for this test is getting start with Asterisk and soft phone. Below is the set up of this test. 
[[File:Asterisk_Internal.png|800px|none|thumb|Internal calls via Dragino]]

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==Install and run Asterisk in Dragino==
make sure your Dragino have network connection, Run:
 root@dragino-bc8622:~# opkg update      // Update the valid package list
 root@dragino-bc8622:~# opkg install asterisk18          // install asterisk v1.8
 root@dragino-bc8622:~# opkg install asterisk18-app-system       // install asterisk system command applications. 

after install Asterisk, we can run:
 root@dragino-bc8622:~# asterisk                        // Start Asterisk Server
 root@dragino-bc8622:~# asterisk -vvvvvvgrc             // Connect to Asterisk Server console, here we can see debug info for asterisk server. 
 Asterisk 1.8.7.1, Copyright (C) 1999 - 2011 Digium, Inc. and others.
 Created by Mark Spencer <markster@digium.com>
 Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
 This is free software, with components licensed under the GNU General Public
 License version 2 and other licenses; you are welcome to redistribute it under
 certain conditions. Type 'core show license' for details.
 =========================================================================
   == Parsing '/etc/asterisk/asterisk.conf':   == Found
 Connected to Asterisk 1.8.7.1 currently running on dragino-bc8622 (pid = 1207)
 Verbosity was 0 and is now 6
    -- Remote UNIX connection
 dragino-bc8622*CLI>
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==Register IP phone extensions==
Set up phone extensions, modify the file /etc/asterisk/sip.conf and add below at the end of this file.
 [dragino-test1]
 type=friend                     ; Friends place calls and receive calls
 context=demo                ; Context for incoming calls from this user
 secret=password
 host=dynamic                    ; This peer register with us
 dtmfmode=rfc2833                ; Choices are inband, rfc2833, or info
 allow=ulaw                      ; dtmfmode=inband only works with ulaw or alaw!

 [dragino-test2]
 type=friend                     ; Friends place calls and receive calls
 context=demo                ; Context for incoming calls from this user
 secret=password
 host=dynamic                    ; This peer register with us
 dtmfmode=rfc2833                ; Choices are inband, rfc2833, or info
 allow=ulaw                      ; dtmfmode=inband only works with ulaw or alaw!

After modify the sip.conf file, we can run below command to active the change
 dragino-bc8622*CLI> core reload
This Dragino has a IP 192.168.1.5. So now the complete resigster infomation for these two accounts are as below:
 Account 1:
 Server IP: 192.168.1.5
 Username: dragino-test1
 Password: password

 Account 2:
 Server IP: 192.168.1.5
 Username: dragino-test2
 Password: password

Now we need to register the Android Mobile to the server, to do this , you can download the free softphone 3CXPhone and install in Android. then create a profile in 3CXphone and input below info:
*Name: Dragino-Test
*User: dragino-test1
*ID: dragino-test1
*Password:  password
*Local PBX IP: 192.168.1.5
Save this profile and the Android phone can register to the Dragino. you should be able to see below info in Dragion after registered:
    -- Registered SIP 'dragino-test1' at 192.168.1.220:33696
       > Saved useragent "3CXPhone for Android 2.0.5" for peer dragino-test1

To make internal calls, we need another extensions, we can set up this in a PC. Download and install the free softphone [http://www.counterpath.com/x-lite-5-for-windows-download.html x-lite] and set up with Account 2 info.

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==Assign Number to extension and make calls==
Now we have two extensions in this small phone system now, we can do it by modifing the /etc/asterisk/extensions.conf file. open this file and find the context '''''[demo]''''', add: 
 exten => 501,1,Dial(SIP/dragino-test1,20,tr)	; number 501 is call to Account 1. 
 exten => 502,1,Dial(SIP/dragino-test2,20,tr)	; number 502 is call to Account 2. 

Then save the change and reload asterisk in asterisk console. 

After doing above, you will be able to make calls between your mobile(phone number:501) and the PC (phone number: 502).

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=Test 2: Control Dragino by dialing numbers=
==Purphose==
Use your mobile to control Dragino by dialling a number. 

==Link system commands with phone numbers==
One way to link numbers and system commands is use the '''''System''''' application in Asterisk. We can add below lines to '''''[demo]''''' context in the /etc/asterisk/extensions.conf file
 exten => 7466,1,System(/bin/echo "test" > /tmp/setpinnum.txt)      -- dial 7466 to call system command
 exten => 7466,2,Hangup                     --hangup the call 

In this link: [[Dragino_SPI#Send_commands_via_SPI_to_AVR_and_control_relay|how to use SPI]]. We show how to control the relay by a linux command. so we can use write lines here to use phone call to control the relay. like below:
 exten => 7467,1,System(lua /root/SPI_Test.lua sh)      --dial 7467 and the system will swith the relay to NO 
 exten => 7467,2,Hangup
 exten => 7468,1,System(lua /root/SPI_Test.lua sl)      --dial 7468 and the system will swith the relay to NC.
 exten => 7468,2,Hangup

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=Test 3: Use IVR (Interactive Voice Response) to control things=
==Purphose==
Use IVR to contorl things, when dial a number, the system will prompt "press 1 to power on device, press 2 to power off". After that, user can power on / off by pressing number 1 or 2.

==Prepare sounds file==
To save storage space on Dragino, the sound file we use are of GSM format. You can convert a .wav, . wma sound file to GSM format by using [http://www.nch.com.au/wavepad/index.html WavePad]. We have prepared three gsm files for our testings:
*press1-on-press2-off.gsm: "press 1 to power on device, press 2 to power off"
*power-on: "Power on device"
*power-off.gsm: "Power off device"

The asterisk default sound directory is '''/var/lib/asterisk/sounds'''. This directory is a temporary directly and all files store here will be erase after reboot. So we upload the sounds file to another directory in Dragino: '''/usr/lib/asterisk/sounds/'''.  and make a soft link to /var/lib/asterisk/sounds to make sure Asterisk can still find sounds after reboot. 
 root@dragino-bc8622:~# mkdir -p /var/lib/asterisk
 root@dragino-bc8622:~# ln -sf /usr/lib/asterisk/sounds/ /var/lib/asterisk/

==Modify dial plan for IVR==
Add a new context in the /etc/asterisk/extensions.conf file:
<source lang="c">
[IVR_TEST]
exten => s,1,Answer()
exten => s,n,Background(press1-on-press2-off)   -- Say file " press1-on-press2-off.gsm"
exten => s,n,WaitExten()                                                -- Wait for user dial a number

exten => 1,1,Playback(power-on)                               -- Say "Power-on"
exten => 1,n,System(lua /root/SPI_Test.lua sh)      --press 1 and the system will swith the relay to NO
exten => 1,n,hangup()
 
exten => 2,1,Playback(power-off)                                --Say "Power-off"
exten => 2,n,System(lua /root/SPI_Test.lua sl)      --press 2 and the system will swith the relay to NC
exten => 2,n,hangup()
</source>



In the file /etc/asterisk/extensions.conf, find the demo() context and add a new dial number:

 <source lang="c">
[demo]
...
...
...
exten => 7469,1, Goto(IVR_TEST,s,1)     -- User Dial 7469 to go to the IVR menu. 
</source>



After modify the extensions.conf file, run "core reload" in Asterisk CLI to reload Asterisk. Then you will be able to reach the IVR menu by dialing '''7469''' in your phone.


=Test 4: Auto dial out on events=
==Purphose==
Show how to use Dragino and Asterisk to auto dial out an extension when there is alarm. 

==Hardware set up==
As shown in below picture, in this test, we use Dragino with [[Temperature_Processing_Module|MT-DS]] to monitor the temperature every 10 seconds. When temperature is higher than our pre-defined upper temperature. Dragino will auto dial to SIP user / dragino-test1 and play back sound file "temperature-too-high.gsm". After the call, if the temperature still too high after 1 minutes. System will call user again. 
[[File:MT-DS_single_sensor.png|600px|none|thumb|Auto dial out test set up]]
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==Modify Dial Plan to promote alarm==
We added a new extension in the [demo] context in /etc/asterisk/extensions.conf. This extension has a number 7470 and it will playback the sound file "temperature-too-high.gsm". 
 exten => 7470,1,Playback(temperature-too-high)
 exten => 7470,n,Hangup

For how to upload the sound files to Dragino, please refer [[Internet_of_Things_and_VoIP#Prepare_sounds_file|Test 3]]

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==Write auto dial out script==
Asterisk has auto dial feature, use this feature, user can auto dial a number base on events. The detail of this feature can be found [http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out here]. In our test, we use simple set up and below is the steps:

First, we need to install the package which support auto dial out. In Dragino, run: 
 root@dragino-bc8622:~# opkg update
 root@dragino-bc8622:~# opkg install asterisk18-pbx-spool
 Installing asterisk18-pbx-spool (1.8.7.1-1) to root...
 Downloading http://downloads.openwrt.org/backfire/10.03.1/atheros/packages/asterisk18-pbx-spool_1.8.7.1-1_atheros.ipk.
 Configuring asterisk18-pbx-spool
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Second, we need to create a script with the auto-dial info. we create a script with file name: '''''outgoing.call''''', This outgoing.call file can be placed in anywhere in the system. 
<source lang="c">
  Channel: SIP/dragino-test1       -- Auto dial to SIP user: dragino-test1
  RetryTime: 60                    
  WaitTime: 30
  Context: demo             -- Context in extensions.conf file
  Extension: 7470           -- extension in extensions.conf file
  Priority: 1               -- Priority to start with in extension. 
</source>
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Third, to active an auto dial, what we need to do is simple copy the file "outgoing.call" to /var/spool/asterisk/outgoing/. We also need to write a script to detect temperature and auto dial user when there is alarm. This script "Temperature_Alarm.lua" do below functions:
*Get the temperature every 10 seconds.
*Print out the temperature
*Check if the temperature exceeds the upper limit. if exceeds, copy file "outgoing.call" to /var/spool/asterisk/outgoing/. Asterisk will auto dial out according to the outgoing.call info. 
*Continue check when there is alarm. if the temperature still too high after 1 minutes, system will auto dial out again. 

"Temperature_Alarm.lua" and "outgoing.call" can be found [http://svn.dragino.com/examples/scripts/ from this link]
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==Run this example==
<syntaxhighlight lang="c" line="GESHI_FANCY_LINE_NUMBERS">
root@dragino-bc8622:~# mkdir -p /var/lib/asterisk                    
root@dragino-bc8622:~# ln -sf /usr/lib/asterisk/sounds/ /var/lib/asterisk/
root@dragino-bc8622:~# mkdir -p /var/spool/asterisk/outgoing/ 
root@dragino-bc8622:~# asterisk
Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk -r' to connect.
root@dragino-bc8622:~# lua Temperature_Alarm.lua 
Current temperature is 21.00
Current temperature is 22.50
Current temperature is 30.50
Temperature is too high! ALARM!    --Alarm here, and auto dial to user. 
Current temperature is 35.00
Temperature is too high! ALARM!
Current temperature is 33.00
Temperature is too high! ALARM!
Current temperature is 31.50
Temperature is too high! ALARM!
Current temperature is 30.00
</syntaxhighlight>
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=Resource=
Examples files about IoT and VoIP can be found [http://svn.dragino.com/examples/IoT_VOIP/ here].

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=Reference=
[http://www.asterisk.org Asterisk official website]

[http://www.voip-info.org Asterisk tutotial website]: Description of each configure file in Asterisk. also lots of examples.